An equalizer by which the overall response characteristics of a reproduction system including a speaker and the like are corrected in accordance with the acoustic characteristics of a listening room is widely used. The acoustic characteristics of a listening room vary depending on the type of room and the installation location of an apparatus for reproducing sound. For example, sound echoes greatly in a wooden-floor room, and sound is absorbed in a bedroom provided with large furniture such as beds. However, sound is hardly absorbed and echoes less in a tatami-floored room provided with no large furniture. Further, the overall acoustic characteristics of a listening room vary between a case where a speaker is placed in parallel with a wall surface of the room and a case where the speaker is placed in a corner of the room. The equalizer corrects output sound with use of an acoustic field control filter so that the quality of the output sound is suited to audiovisual environments having different acoustic characteristics.
For example, as a conventional technique for correcting the overall response characteristics of a reproduction system by adjusting audio quality, Patent Document 1 discloses an acoustic characteristic correction apparatus that allows a user to easily set a desired response characteristic of the reproduction system as a preferred characteristic.
The following describes the acoustic characteristic correction apparatus of Patent Document 1 more in detail. FIGS. 14(a) through 14(e) show various types of characteristics obtained in steps taken by the acoustic characteristic correction apparatus of Patent Document 1 in correcting acoustic characteristics. First, the acoustic characteristic correction apparatus of Patent Document 1 reproduces a measuring signal such as a band signal or a TSP signal with use of a speaker included in a reproduction system that is to be corrected, collects the reproduced sound with use of a microphone, and then calculates the response characteristics, i.e., measured characteristics (see FIG. 14(a)) of the reproduction system. Next, the acoustic characteristic correction apparatus calculates, as a correction characteristic (see FIG. 14(c)), a difference between the preferred characteristic (see FIG. 14(b)) set by the user and the measured characteristics, and then makes a modification as needed. Furthermore, the acoustic characteristic correction apparatus calculates corresponding impulse responses (see FIG. 14(d)) by performing inverse Fourier transform of the determined correction characteristic, and sets, as coefficients of an equalizer (FIR (finite impulse response)) filter, level values in respective positions of the calculated impulse responses on the time axis.
Patent Document 1 describes, as a method for calculating impulse responses from a correction characteristic by inverse Fourier transform, an embodiment that employs linear-phase inverse Fourier transform.
According to the linear-phase inverse Fourier transform, impulse responses are calculated by dividing the corrected characteristic into bands, by calculating a power average for each of the bands, by interpolating the power average values by spline interpolation or the like into 4096 pieces of data that can be subjected to Fourier transform, and then by performing inverse Fourier transformation of complex format data having a real part in which the interpolated data have been set (and an imaginary part that has been entirely set to 0). It should be noted here that the real part of the complex format data corresponds to an amplitude term and the imaginary part of the complex format data corresponds to a phase term. Moreover, since that imaginary part of the complex format data which corresponds to a phase term has been entirely set to 0 as described above, the impulse responses calculated by the linear-phase inverse Fourier transform contain no phase information.
Since a filter calculated by the linear-phase inverse Fourier transform, i.e., a linear-phase filter contains no phase information, filter coefficients are easily calculated, and a good frequency transfer characteristic is obtained. However, this makes it impossible to correct a phase lag caused by the reproduction system.
In order to solve this problem, there is a technique for correcting the acoustic characteristics of a reproduction system by using an inverted filter containing phase information. Non-patent Document 1 describes a method for designing the inverted filter.
The following provides an outline of the inverted filter. The inverted filter H(z) is represented by H(z)=1/C(z), where C(z) is the transfer characteristic of the reproduction system. This formula indicates that the introduction of the inverted filter H(z) equalizes an output and input of the reproduction system. That is, the inverted filter H(z) is designed so that impulse responses of the reproduction system form a unit impulse (delta function δ(n)). However, a normal reproduction system is not a minimum-phase transition system and contains a propagation delay. Therefore, the inverted filter H(z) is designed so that the impulse responses are changed to form δ(n−M), where M is referred to as a modeling delay.
Further, depending on the transfer characteristic of the reproduction system, H(z)=1/C(z) cannot be directly solved. However, an approximation of the inverted filter can be calculated, for example, in accordance with the least squares principle. The inverted filter designed in accordance with the least squares principle is generalized as H(z)=C*(z)/C*(z)C(z), where C(z) is a complex number and C*(z) is a conjugate complex number of C(z).
Other various techniques have been proposed as a technique for correcting response characteristics by using an acoustic field control filter. For example, Patent Document 2 discloses an amplification articulation improving device capable of realizing amplification with high articulation in an environment where reverberations are likely to be heard. The following describes the amplification articulation improving device of Patent Document 2 more in detail. FIG. 15(a) shows the flow of a process by which the amplification articulation improving device of Patent Document 2 improves the articulation of amplification. As shown in FIG. 15(a), the amplification articulation improving device of Patent Document 2 measures an impulse response in a closed space and determines for each 1/n band whether or not the reverberation time exceeds a predetermined period of time. In cases where the reverberation time exceeds the predetermined period of time, the amplification articulation improving device calculates difference energy between the measured impulse response and an impulse response calculated from direct sound, and stacks the calculated difference energy in a memory. FIG. 15(b) shows difference energy for each 1/n (octave) frequency band. Furthermore, after the process of determining reverberation time and stacking difference energy has been performed for each 1/n frequency band, an inverse transfer function is calculated in accordance with the difference energy calculated for each frequency band, and equalizer parameters that satisfy the transfer function are set in a filter. This enables the amplification articulation improving device of Patent Document 2 to reduce the sound volume level of a frequency band having such a long reverberation time as to affect articulation. This makes it possible to realize amplification with high articulation without causing a big change in original audio quality.
Incidentally, a FIR filter is represented as an arrangement in which an output is obtained by causing a delay element (buffer) to sequentially delay input data, by causing a multiplier to multiply filter coefficients preset in the delay outputs, and by causing an adder to add the multiplied outputs. That is, the FIR filter processes a signal by performing a product-sum computation process. In order to realize a high-order FIR filter, it is necessary to perform such a product-sum computation process a large number of times. Moreover, in causing the FIR filter to process a signal, a DSP (digital signal processor) capable of performing multiplication and addition in one machine cycle and processing a product-sum computation at a high speed is used.
The FIR filter performs a product-sum computation of convolution as expressed by the following formula:y(n)=h0·x(n)+h1·x(n−1)+h2·x(n−2)+ . . . +hN·x(n−N)
where y(n) is an output signal value, x(n−i) (i=0, 1, . . . N) is a present or past input signal value, and hi (i=0, 1, . . . N) is a filter coefficient (weight). That is, the output signal value of the FIR filter is represented by an average weighted with the present or past input signal value.
It should be noted that the FIR filter includes taps (each of which is a block constituted by the aforementioned delay element, the aforementioned multiplier, and the aforementioned adder) whose number corresponds to the number of terms of hi·x(n−i) included in the foregoing formula. Moreover, the characteristics of FIR filter are changed by changing the number of taps constituting the filter and by changing the value of hi of each of the taps. The larger the number of taps is, the higher the resolution of the frequency is. This results in higher performance of the filter.
However, an increase in the number of taps of the FIR filter (i.e., the number of filter coefficients) causes an increase in the number of such product-sum computations as described above, thereby causing an increase in the number of processes to be performed by the DSP. This makes it necessary to use a high-performance DSP, thereby causing an increase in cost necessary for constituting the FIR filter. Therefore, it is necessary to consider a trade-off between performance and cost in selecting a DSP that is to be mounted on a product.
[Patent Document 1]
Japanese Unexamined Patent Application Publication No. 327089/1994 (Tokukaihei 6-327089; published on Nov. 25, 1994)
[Patent Document 2]
Japanese Unexamined Patent Application Publication No. 224898/2003 (Tokukai 2003-224898; published on Aug. 8, 2003)
[Non-patent Document 1]
http://www.sound.sie.dendai.ac.jp/dsp/Text/PDF/C hap7-2.pdf (confirmed on Jan. 25, 2007)
As described above, a DSP to be mounted on a product is selected in consideration of a trade-off between performance and cost. Moreover, a FIR filter is designed in consideration of the capability of the selected DSP to perform a product-sum computation. Therefore, the number of taps of the FIR filter (i.e., the number of filter coefficients) is limited depending on the specifications of the DSP.
In cases where the filter coefficients of the FIR filter are calculated by the aforementioned inverted filter, first, impulse responses are measured with use of a TSP method or the like in a reproduction system whose audio quality is to be corrected, and a frequency characteristic of the impulse responses thus measured (hereinafter referred to as “measured impulse responses”) is calculated. Then, a frequency characteristic of the inverted filter is calculated in accordance with the frequency characteristic thus calculated, and impulse responses corresponding to the inverted filter (such an impulse response being hereinafter referred to as “inverted filter impulse responses”) are calculated by performing inverse Fourier transform of the frequency characteristic of the inverted filter. The inverted filter impulse responses are set as the filter coefficients of the FIR filter.
It should be noted that the aforementioned process of calculating the coefficients of the FIR filter is digital signal processing. After the measured impulse responses are loaded as a continuous analog signal, the signal is sampled so as to be converted into discrete digital signals. At this time, in order that high frequency component information contained in the original analog signal is incorporated into the digital signals, it is necessary to sufficiently narrow each sampling interval, i.e., to sufficiently increase the number of samples. Then, data (i.e., filter coefficients of the FIR filter) representing the aforementioned inverted filter impulse responses are calculated in accordance with data representing the measured impulse responses thus sampled.
At this time, the number of pieces of calculated data that represent the inverted filter impulse responses is identical to the number of pieces of data that represent the measured impulse responses. Then, the calculated data representing the inverted filter impulse responses are set as the coefficients of the FIR filter. However, as described above, the number of taps of a FIR filter (i.e., the number of filter coefficients) is limited depending on the specifications of a DSP. Therefore, all the calculated data representing the inverted filter impulse responses cannot be used as the coefficients of the FIR filter. Thus, the inverted filter impulse responses are clipped. That is, only a part of the calculated data representing the inverted filter impulse responses is taken out as the coefficients of the FIR filter.
However, in cases where only a part of the data representing the inverted filter impulse responses is set as the coefficients of the FIR filter, data that are not set as coefficients are discarded. This causes deterioration in performance of the FIR filter. Therefore, the correction of audio quality with use of the FIR filter thus calculated causes a serious error in corrected impulse responses, thereby causing a gain difference in a gain-frequency characteristic of the corrected impulse responses.
FIG. 16 shows inverted filter impulse responses calculated in accordance with measured impulse responses (the number of measured impulse responses sampled: 512). The number of pieces of data that represent the inverted filter impulse responses of FIG. 16 is 512, which is identical to the number of measured impulse responses sampled. In cases where the number of taps of a FIR filter is limited to 256 by the specifications of a DSP, for example, 256 pieces of data centered around the peak value of amplitude are extracted from the inverted filter impulse responses of FIG. 16 as coefficients of the FIR filter. That is, the pieces of data that fall within a range surrounded by the dashed line of FIG. 16 are discarded. In this case, the amplitude of the range of impulse responses surrounded by the dashed line of FIG. 16 is great, and is not small enough to be ignored as compared with the amplitudes of the whole impulse responses. Therefore, even if the audio quality is corrected by the FIR filter thus calculated, the corrected impulse responses and the corresponding frequency characteristic contain a large number of errors.
The present invention has been made in view of the foregoing problems, and it is an object of the present invention to provide a filter coefficient calculation device, a filter coefficient calculation method, a control program, a computer-readable storage medium, and an audio signal processing apparatus, each of which makes it possible to correct acoustic characteristics with high precision even in cases where the number of filter taps is limited.